New PeerConnection handler in Chrome to support latest PeerConnection draft

content/content_tests.gypi

 # Copyright (c) 2012 The Chromium Authors. All rights reserved.
 # Use of this source code is governed by a BSD-style license that can be
 # found in the LICENSE file.
 
 {
   'targets': [
     {
       'target_name': 'test_support_content',
       'type': 'static_library',
       'defines!': ['CONTENT_IMPLEMENTATION'],
@@ skipping 135 matching lines @@
         }],
         ['enable_webrtc==1', {
           'sources': [
             'renderer/media/mock_media_stream_dependency_factory.cc',
             'renderer/media/mock_media_stream_dependency_factory.h',
             'renderer/media/mock_media_stream_dispatcher.cc',
             'renderer/media/mock_media_stream_dispatcher.h',
             'renderer/media/mock_peer_connection_impl.cc',
             'renderer/media/mock_peer_connection_impl.h',
             'renderer/media/mock_web_peer_connection_00_handler_client.cc',
-            'renderer/media/mock_web_peer_connection_00_handler_client.h',
-            'renderer/media/mock_web_peer_connection_handler_client.cc',
-            'renderer/media/mock_web_peer_connection_handler_client.h',
+            'renderer/media/mock_web_peer_connection_00_handler_client.h',            

[wjia(left Chromium), 2012/08/12 17:06:52]

remove extra spaces at the end.

[perkj_chrome, 2012/08/13 07:35:46]

Done.

+            'renderer/media/mock_web_rtc_peer_connection_handler_client.cc',
+            'renderer/media/mock_web_rtc_peer_connection_handler_client.h',
             'test/webrtc_audio_device_test.cc',
             'test/webrtc_audio_device_test.h',
           ],
           'dependencies': [
             '../third_party/libjingle/libjingle.gyp:libjingle_peerconnection',
             '../third_party/webrtc/modules/modules.gyp:audio_device',
             '../third_party/webrtc/modules/modules.gyp:video_capture_module',
             '../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
             '../third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
             '../third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine_core'],
@@ skipping 264 matching lines @@
       'conditions': [
         ['enable_webrtc==1', {
           'sources': [
             'browser/renderer_host/p2p/socket_host_test_utils.h',
             'browser/renderer_host/p2p/socket_host_tcp_unittest.cc',
             'browser/renderer_host/p2p/socket_host_tcp_server_unittest.cc',
             'browser/renderer_host/p2p/socket_host_udp_unittest.cc',
             'renderer/media/media_stream_dispatcher_unittest.cc',
             'renderer/media/media_stream_impl_unittest.cc',
             'renderer/media/peer_connection_handler_jsep_unittest.cc',
+            'renderer/media/rtc_peer_connection_handler_unittest.cc',
             'renderer/media/rtc_video_decoder_unittest.cc',
             'renderer/media/webrtc_audio_device_unittest.cc',
             'renderer/p2p/p2p_transport_impl_unittest.cc',
           ],
           'dependencies': [
             '../third_party/libjingle/libjingle.gyp:libjingle_peerconnection',
             '../third_party/webrtc/modules/modules.gyp:video_capture_module',
             '../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
             '../third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
             '../third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine_core',
@@ skipping 373 matching lines @@
                 '-buildfile',
                 '<(DEPTH)/content/shell/android/javatests/content_shell_test_apk.xml',
               ]
             }
           ],
         },
       ],
     }],
   ],
 }